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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 2ba95bec7593705962f8da9ea23ef457e4c23c48
  1. 2ba95be Upgrade scoped_ptr to Chromium's latest version. by andrew@webrtc.org · 11 years ago
  2. e5fd264 Roll chromium_revision 228675:229708 by kjellander@webrtc.org · 11 years ago
  3. a597fc2 WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN. by andrew@webrtc.org · 11 years ago
  4. 4535225 Add CurrentLayerId() to temporal layers. by marpan@webrtc.org · 11 years ago
  5. 1e1938a Updated WebRTC version to 3.45 by elham@webrtc.org · 11 years ago
  6. 61e533c Framework for testing bandwidth estimation. by solenberg@webrtc.org · 11 years ago
  7. 97d0fc6 Changing the bitrate clamping in BitrateControllerImpl by henrik.lundin@webrtc.org · 11 years ago
  8. 0c6fa57 AutoMute: Adding channel_id parameter to callback. by henrik.lundin@webrtc.org · 11 years ago
  9. 3ba57eb Implement I420FrameCallbacks in Call. by pbos@webrtc.org · 11 years ago
  10. dea5a74 Make sure the first frame isn't dropped. by pbos@webrtc.org · 11 years ago
  11. 05ee189 Move audio_e2e_harness into include_tests==1 condition. by kjellander@webrtc.org · 11 years ago
  12. 8ee76b9 Add audio_e2e_test target to tools.gyp by kjellander@webrtc.org · 11 years ago
  13. 6c0739e Protect _transportPtr, which can be accessed by different threads in the case of external transport. This change avoid the potential use-after-free, e.g. the case in the reported bug. by wu@webrtc.org · 11 years ago
  14. 9b307a7 Have padding decay to zero if no frames are being captured. by stefan@webrtc.org · 11 years ago
  15. 6be4250 Disable the -Wno-unused-const-variable Clang warning on Mac by kjellander@webrtc.org · 11 years ago
  16. c4579f3 Minor comment fix after clang reformat. by andrew@webrtc.org · 11 years ago
  17. b746a33 MouseCursorMonitor implementation for OSX and Windows. by sergeyu@chromium.org · 11 years ago
  18. 224c0f5 Fix tsan failures in channel.cc regarding to the volume settings. by wu@webrtc.org · 11 years ago
  19. 1e6493d Check the number of playout channels instead of the send channels in StopPlayout() by xians@webrtc.org · 11 years ago
  20. b1ef0d7 Compound/reduced-size RTCP in VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  21. de16548 Remove unused kPowTableFrac which causes anroid clang build failure. by wu@webrtc.org · 11 years ago
  22. 6133dd5 Fixed issue with how MTU is calculated. by sprang@webrtc.org · 11 years ago
  23. e44b42d Don't pad if only one stream is sent, except if auto muted. by stefan@webrtc.org · 11 years ago
  24. a19dab9 Revert "Disable tests for TSan v2" by kjellander@webrtc.org · 11 years ago
  25. 4fe8543 Wired up max packet size and added simple test. by sprang@webrtc.org · 11 years ago
  26. 55ca27e Run FullStack tests without render windows. by pbos@webrtc.org · 11 years ago
  27. 1c83344 Remove TSan v2 disabled test in condition_variable_unittest.cc by kjellander@webrtc.org · 11 years ago
  28. 976adc0 Open file in binary in CreateFromYuvFile(). by pbos@webrtc.org · 11 years ago
  29. 28f6166 Add MouseCursorRenderer. by sergeyu@chromium.org · 11 years ago
  30. 92da5d7 Add MouseCursorCapturer interface with implementation for X11. by sergeyu@chromium.org · 11 years ago
  31. 0f281aa Roll chromium_revision 226126:228675 and fix clang warnings by kjellander@webrtc.org · 11 years ago
  32. 7865560 Make RtpData and RtpFeedback destructors public. by stefan@webrtc.org · 11 years ago
  33. 2390091 Move ChromaGenerator to common_video/. by pbos@webrtc.org · 11 years ago
  34. 8dc840d Compile out unused kMinTrustedDelayMs. by andrew@webrtc.org · 11 years ago
  35. c1b7718 Android: Fixes WebRTCDemo build (missing Java code). by henrike@webrtc.org · 11 years ago
  36. 53fa5da NetEq4: Removing templatization for AudioVector by henrik.lundin@webrtc.org · 11 years ago
  37. f24a93f Remove empty line in SharedXDisplay::RemoveEventHandler. by sergeyu@chromium.org · 11 years ago
  38. a171eae Android OpenSlDemo: remove some usages of deprecated APIs that is breaking the bots. by henrike@webrtc.org · 11 years ago
  39. 5e8b020 Add event handling in SharedXDisplay. by sergeyu@chromium.org · 11 years ago
  40. a6295d3 Add DesktopCaptureOptions class. by sergeyu@chromium.org · 11 years ago
  41. 4b795a1 WebRTCDemo: Fixes warning for devices with pre-17 API level. Also fixes broken build build.xml and project.properties. by henrike@webrtc.org · 11 years ago
  42. 0b7aefe Reorganize GYP targets to make webrtc.gyp more usable. by kjellander@webrtc.org · 11 years ago
  43. f50f118 clang-format audio_processing/aec/* by andrew@webrtc.org · 11 years ago
  44. fd03cb1 Add a parameter to audioproc for overriding the delay. by andrew@webrtc.org · 11 years ago
  45. 472d9a7 Updated WebRTC version to 3.44 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
  46. 96ea7ac Revert r4934 "Add a tool for parsing an RTP file and outputting the BWE relevant fields." by stefan@webrtc.org · 11 years ago
  47. 2dd26d8 Fix build error in r4934. by stefan@webrtc.org · 11 years ago
  48. 6dea08f Add a tool for parsing an RTP file and outputting the BWE relevant fields. by stefan@webrtc.org · 11 years ago
  49. e3976cf Wrap ACM2 code inside acm2 namespace. This gurantees that one ACM would not use components of others by accident. by turaj@webrtc.org · 11 years ago
  50. e9a3119 Accounting for wrap-around of timestamps. by turaj@webrtc.org · 11 years ago
  51. 7a4ff8a VPM: Fixing namespace by mikhal@webrtc.org · 11 years ago
  52. d4b124a Android: enable camera video stabilization when available. by fischman@webrtc.org · 11 years ago
  53. 7de0054 Add owners to [webrtc,talk]/build and *.isolate (take 2) by kjellander@webrtc.org · 11 years ago
  54. b7768d5 Remove unused Android dummy APK by kjellander@webrtc.org · 11 years ago
  55. 9670be6 Add APK and isolate target for video_engine_tests by kjellander@webrtc.org · 11 years ago
  56. 28ea6f8 Clean up AudioProcessing defaults and errors. by andrew@webrtc.org · 11 years ago
  57. 9466714 Add owners to [webrtc,talk]/build and *.isolate by kjellander@webrtc.org · 11 years ago
  58. d64f84f Only declare kDelayDiffOffset when used. by andrew@webrtc.org · 11 years ago
  59. a299658 Unbreaks Android build after r4915. by henrike@webrtc.org · 11 years ago
  60. b17cc30 Revert r4913 that reverts r4911. Original CL description: by andresp@webrtc.org · 11 years ago
  61. f327500 Android standalone: remove some usages of deprecated APIs and prevent further regressions. by fischman@webrtc.org · 11 years ago
  62. a5c9463 VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android. by fischman@webrtc.org · 11 years ago
  63. cca73e7 Revert 4911 "Adding temporal layer strategy that keeps base laye..." by turaj@webrtc.org · 11 years ago
  64. 0a202c7 Reformatting VPM: First step - No functional changes. by mikhal@webrtc.org · 11 years ago
  65. 05927b8 Adding temporal layer strategy that keeps base layer framerate at an acceptable value. by andresp@webrtc.org · 11 years ago
  66. 886983e Minor fix to avoid breakage by henrik.lundin@webrtc.org · 11 years ago
  67. 8217db9 Disable Receiver unittests on Android. by turaj@webrtc.org · 11 years ago
  68. 0d6ee5d ACM test are modified to run with both ACM1 and ACM2. by turaj@webrtc.org · 11 years ago
  69. 3de1b22 Fix include of isolate.gypi by kjellander@webrtc.org · 11 years ago
  70. c73a60e Android OpenSL: Fixes faulty assertion in jni-code. by henrike@webrtc.org · 11 years ago
  71. 15eda06 Remove ReturnTrace from DeregisterCallback(). by pbos@webrtc.org · 11 years ago
  72. f9159ec Remove templatization of the AudioVector test by henrik.lundin@webrtc.org · 11 years ago
  73. 712d30f Workaround issue with stdin on Windows. by kjellander@webrtc.org · 11 years ago
  74. d6c8fec APK for opensl loopback. by henrike@webrtc.org · 11 years ago
  75. 8ee2137 Implement TraceCallbacks in Call. by pbos@webrtc.org · 11 years ago
  76. 55312fe Piping AutoMuter interface through to ViE API by henrik.lundin@webrtc.org · 11 years ago
  77. adba142 Added support for sending and receiving RTCP XR packets: by asapersson@webrtc.org · 11 years ago
  78. c6c6c1d Stop timer in ~EventWindows(). by pbos@webrtc.org · 11 years ago
  79. 37ae275 Update sampling rate and number of channels of NetEq4 if decoder is changed. by turaj@webrtc.org · 11 years ago
  80. be46543 Test multiple send/receive streams in Call. by pbos@webrtc.org · 11 years ago
  81. 7befa0c Remove include_dirs from utility. by pbos@webrtc.org · 11 years ago
  82. d2ca96e PeerConnection(Android): enable tracing to logcat. by fischman@webrtc.org · 11 years ago
  83. cf71152 Reset audio bufer if codec changes, b/10835525. by turaj@webrtc.org · 11 years ago
  84. 324dac6 Ensure adjusted "known delay" doesn't drop below zero. by andrew@webrtc.org · 11 years ago
  85. 80f29d8 NetEq4: Removing templatization for AudioMultiVector by henrik.lundin@webrtc.org · 11 years ago
  86. d069ddf Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
  87. 9d8bbb7 Remove include_dirs from video_render. by pbos@webrtc.org · 11 years ago
  88. 6ea626e Remove include_dirs from video_capture. by pbos@webrtc.org · 11 years ago
  89. 5c9d8ee Revert 4876 "Support for CELT in NetEq4." by tina.legrand@webrtc.org · 11 years ago
  90. 38b1789 Propagate AutoMuter interface out to VideoCodingModule by henrik.lundin@webrtc.org · 11 years ago
  91. 1364cf1 1. adding request of ACM version in the manual mode of voe_auto_test by minyue@webrtc.org · 11 years ago
  92. 10b35b2 Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
  93. 197a5eb Change the parameters of calculating maximum decode time. by wuchengli@chromium.org · 11 years ago
  94. 7540ded Makes OpensSL default audio implementation/device on Android. by henrike@webrtc.org · 11 years ago
  95. aaa1ce6 Add protection to few more methods of AudioDeviceLinuxALSA. Those methods can be called from by wu@webrtc.org · 11 years ago
  96. db10d03 Only use -lm on Linux in ISAC. by andrew@webrtc.org · 11 years ago
  97. 45f59dd Remove test parameters from CallTest. by pbos@webrtc.org · 11 years ago
  98. 8ac5bf4 With ACM2 and NetEq4, VoE fuzz test very often fails. by minyue@webrtc.org · 11 years ago
  99. 1a58624 Remove include_dirs from tools. by pbos@webrtc.org · 11 years ago
  100. 109108e Remove include_dirs from test. by pbos@webrtc.org · 11 years ago